FIG. 1 is a schematic diagram showing the architecture of a wireless communication network. As shown in FIG. 1, a typical wireless communication network generally includes one or more terminals, base stations and a core network, wherein different base stations may connect with each other by requirements, and a network composed of the base stations, which is usually named as a RAN (radio access network), is responsible for access layer transactions such as radio resource management. The radio resources including uplink and downlink resources are scheduled by the base stations according to the manner of sharing resources. Base stations are all connected to the core network, and specifically each of the base stations can be connected to one or more core network nodes (CN). The core network is responsible for non-access layer transactions, such as billing and location management. The terminal may be a device communicating with the network, such as a mobile phone or a laptop computer.
In the wireless communication system which adopts packet transmission mode, such as LTE or LTE-A system, uplink and downlink data are generally transmitted with dynamic scheduling.
The following will make descriptions with a VoIP service as an example.
Currently, the VoIP service is a very important service and basic principles of the VoIP service include: at a transmitter, compressing voice data codes using a voice compressing algorithm, packing the compressed voice data according to the TCP/IP protocol, and then transmitting the packed voice data to a receiver through an IP network; and at a receiver, concatenating the packed voice data and de-compressing to restore them into the original voice data, in order to transmit the voice data over the Internet.
Such VoIP service characterizes in constant rate and relatively small size VoIP packet. As shown in FIG. 2, the VoIP service includes two parts: talkspurt and silent period. During the talkspurt, a voice encoder generates a VoIP packet every 20 ms; whereas during the silent period, the voice encoder generates a SID (silence indicator) packet every 160 ms.
Since the size of a VoIP packet is very small and a large amount of signaling is needed to notify the VoIP user when the base station performs resource allocation and scheduling, it will severely limit the number of VoIP users in the system. Therefore, a more effective scheduling scheme is needed for the VoIP service.
In related art, as shown in FIG. 3, since the size of the VoIP packet and arriving intervals of the VoIP packets are well regulated, a persistent scheduling scheme is adopted to pre-allocate a fixed location and fix-sized radio resources for the VoIP service in order to reduce overhead of control signaling, and a modulation coding scheme (MCS) is specified for transmitting the VoIP packets. However, since HARQ and silence suppression techniques are adopted, the original regulation of VoIP data transmission may be influenced, so dynamic scheduling of control signaling is needed as compensation for the persistent scheduling scheme, which is referred to as a semi-persistent scheduling scheme in the LTE or LTE-A system.
The semi-persistent scheduling scheme is a semi-dynamic scheduling algorithm based on the talkspurt, i.e. the VoIP user occupies resources during the talkspurt and releases the occupied resources during the silent period. Characteristics of the semi-persistent scheduling are as follows.
The initial transmission of the VoIP packets conforms to the persistent scheduling scheme; for retransmission of the VoIP packets, the downlink adopts the dynamic resource allocation scheme, while the uplink adopts the dynamic or persistent scheduling scheme; if dynamic signaling for the dynamic resource allocation exists during the initial transmission of the VoIP packets, the dynamic resource allocation scheme is adopted.
Herein, the initial transmission of the VoIP packets conforms to the persistent scheduling scheme, i.e. the resources adopted by the VoIP packets conform to a certain regulation, while this regulation is notified by the base station to the VoIP user when a VoIP connection is established and the base station per-allocates resources for the VoIP packets. In this case, the VoIP user will exactly know his/her own resource units every 20 ms, without the signaling notification. The resources within an interval (19 ms) between two VoIP packets are managed uniformly by the base station and can be allocated to any service or user.
In addition, the VoIP packet which is transported from the application layer is complementary with a RTP/UDP/IP packet header after passing the network layer. The size of the packet header is generally 40/60 bytes. Since the size of the VoIP packet is comparatively small and smaller than the packet header, the VoIP header needs to be compressed. In related art, the RoHC (Robust Header Compression) technique is adopted to compress the header of the VoIP packet in the PDCP (Packet Data Convergence Protocol) layer. According to the RoHC technique, the header with the size of 40/60 bytes can be compressed to 3˜15 bytes. Thus, the size of the VoIP packet arriving at the MAC/PHY layer is variable and therefore the resource requirements of different size VoIP packets are different, as shown in Table 1.
TABLE 1AMRcodecCompressedTDD DLTDD ULFDD DLFDD ULmode(bytes)(PRBs)(PRBs)(PRBs)(PRBs)AMR17~292~32~31~21~24.75 kbpsAMR18~302~32~31~21~25.15 kbpsAMR19~312~32~31~21~25.90 kbpsAMR21~332~32~3226.70 kbpsAMR23~352~32~3227.40 kbpsAMR25~372~32~3227.95 kbpsAMR30~423~43~42~32~310.2 kbpsAMR35~473~43~42~32~312.2 kbpsAMR SID10~221~21~21~21~2
As shown in Table 1, the MCS adopts QPSK2/3, and when the AMR (Adaptive Multi Rate) coding algorithm adopts 12.2 kbps, the size of the compressed VoIP packet is 35˜47 bytes. In this case, the TDD DL (downlink) and UL (uplink) occupies 3˜4 RBs (resource blocks) in one subframe, and the FDD DL and UL occupies 2˜3 RBs in one subframe.
FIG. 4 is a schematic diagram showing the division of a comparatively large VoIP packet into continuous packets. As shown in FIG. 4, when a packet X with normal size is transmitted, pre-allocated resource Y can meet the requirement of the VoIP packet transmission. However, when a larger packet X+1 is transmitted, the pre-allocated resource Y cannot meet the requirement of the larger VoIP packet transmission, and thus only a part of the VoIP packet (the part without a shade) can be transmitted while the other part A (the part with shade) cannot be transmitted. Thus, the part A of the packet needs to occupy part of resources Y1 of packet X+2 and the part B of the packet X+2 needs to occupy part of resources Y2 of packet X+3. Therefore, when a comparatively large VoIP packet is transmitted, this packet will be divided and this kind of dividing will produce a knock-on effect and subsequent VoIP packets will all need to be divided. Therefore, the transmission of the VoIP packet will have delay and a new header is produced.
Currently there are several solutions to solve the problem that occurs when transmitting variable size VoIP packets, and the solutions are described as follows.
Solution one: the pre-allocated resource is set according to the size of the largest VoIP packet; the problems with this solution are that a large amount of resource is wasted and the number of accommodated users is small.
Solution two: the pre-allocated resource is set according to the size of the smallest VoIP packet and when a larger VoIP packet is to be transmitted, the dynamic resource allocation scheme is adopted; the problem with this solution is that more physical layer control signaling (PDCCH) is needed.
Solution three: the pre-allocated resource is set according to the size of the smallest VoIP packet and when a larger VoIP packet is to be transmitted, the comparatively high MCS is adopted; the problem with this solution is that a comparatively large PER (packet error rate) is induced, the retransmission times are increased and the delay is increased.
Solution four: the pre-allocated resource is set according to the size of the largest VoIP packet and when a smaller VoIP packet is to be transmitted, the dynamic resource allocation scheme is adopted and the remaining part of the pre-allocated resource for this VoIP user is allocated to other services or users; the problem with this solution is that the number of accommodated users is comparatively small.
Chinese patent application No. 200710151704.9, with publication No. CN 101127806A, discloses a downlink voice IP service scheduling method, the entirety of which is incorporated herein by reference. According to this method, scheduling overhead can be reduced to some extent, but the problem occurs when transmitting variable size VoIP packets still exists.
US patent publication No. US2008/0062944 A1, entitled “Apparatus and Method for Automatic Repeat Request Signaling with Reduced Retransmission Indications in a Wireless VoIP Communication System”, is incorporated herein by reference in its entirety.
US patent publication No. US2008/0062178 A1, entitled “VOIP GROUP RESOURCE MANAGEMENT”, is incorporated herein by reference in its entirety.
US patent publication No. US2008/0025337 A1, entitled “Apparatus and Method for Handling Control Channel Reception/Decoding Failure in a Wireless VoIP Communication System”, is incorporated herein by reference in its entirety.
Although the above U.S. patent applications disclose scheduling schemes for VoIP, they adopt BITMAP and packet scheduling schemes, but not the persistent scheduling scheme, and they fail to solve the problem occurs when transmitting variable size VoIP packets yet.